similar to: Automatic Speech Recognition and Text To Speech using iSpeech

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

Hello, FastAGI doesn't get (or deserve) much love these days but a lot of people are still widely using it. Here is a small project of mine, started trying to scratch my own itch, that some might find useful. https://github.com/zaf/agitator It is a reverse proxy for the FastAGI protocol with some interesting features. Most noticeably request based routing, HA/failover, load balancing and

Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called "Microsoft Translator". This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Tue, 2016-02-23 at 17:06 +0000, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F.

I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ It's similar to app_festival and app_flite. It's only tested against asterisk 1.6.1 on x86 Linux but it must be working for other 1.6 branches too. Comments, fixes and suggestion are

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Hi Daniel On Mon, 2016-02-22 at 19:40 +0100, Daniel Heckl wrote: > try this http://zaf.github.io/asterisk-speech-recog/. > I have tested it myself, it works very well. I wanted to try it, but I obtain the following error message: "speech-recog.agi,en-US: API key is missing. Aborting. " :-( What am I missing? Any hint? Frank

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

Hi Marcelo, Thanks for replying, I do not know what this branch is. Could you please let me know. Also, I enabled google cloud speech API only from the console. Do I need more API enabled? On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhterres at gmail.com> wrote: > Did you already tried the cloud_api branch? > > Regards, > > Marcelo H. Terres <mhterres at

Hi, I'm trying to integrate Google cloud speech recognition v2 in it. I can get the audio recorded, have created Service key and API key but whenever I try to access it, I just get 403 access denied. I am at my wits end here. Has anybody tried it ? were you successful ? Could you please guide me how to do it ? I'll be grateful to you if this works ! -- Warm Regds. MathuRahul

I am not aware of any alive project on this field. My client is interested in it. He told me that Google Translate web service used to provide pronunciation of hebrew texts, but now speech button is disabled for Hebrew, and also direct requesting for speech generation (by Lefteris Zafiris's awsome scripts) fails. I can work on Asterisk integration by myself, i'd be happy to know of such

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, First, as far as I can tell, this topic hasn't been mentioned in the mailing lists in awhile (atleast not as far as google can tell). Is there any client/program that can send "net start" requests to windows boxes. The ability to start a program as a service on a windows box remotely would obviously be very handy. I am aware of

Hi Jonathan Thanks ! That would indeed be wonderful, at this point I really do not care whether I need to use Python or Lua or JS. I was following http://zaf.github.io/asterisk-speech-recog/ but hit a road end with (for the lack of sane word ) copulating Google's Key On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconcepts at gmail.com> wrote: > Yes! But I can only tell you if